There is a Neumann one that matrixes at mic level, used for the RSM191. As far as line level matrix boxes, AEA used to have one, but it appears discontinued now.

http://www.wesdooley.com/aea/ds_MS38_MarkII.html

This is something a half decent electronics tech could put together for you. I got mine to make me a little MS matrix pre with Lundahl's in it, its a great box.

Scott Sedillo (Zilla) reads this board, I am sure he would do this in his sleep.

Yes, I ran into this problem, but are you sure you mean encoder and not decoder? A decoder takes the output from a pair of microphones in M-S configuration and makes a stereo L-R pair of channels from them. An encoder is not often found except in front of a decoder for special uses.

There is a transformer-based decoder available from Sowter: http://www.sowter.co.uk/specs/8379.htm

Frustrated by the lack of a commercial electronic M-S unit, I designed a line-level encoder/decoder. It can be used stand-alone or as an insert into an M-S channel pair to convert them to L-R, or as a width control in a standard L-R channel pair. It worked sufficiently well that an audio company is currently interested in producing it commercially. I'm well aware that this isn't a commercial forum, but if it's of interest, I can give further details of the unit when available.

I would vouch for the simplicity of a transformer-based solution, but there is one thing to keep in mind - you can't pass phantom power through a transformer. So, if you intend to use a transformer-based MS decoder on the input of a mic preamp (to decode before recording) it will be a problem for condensers unless they run on batteries. An MS rig using ribbons wouldn't be a problem, and you may be able to steal a bit of noise-free voltage gain from the transformers at the same time.

No problems at line level, however...

BTW, if you have an MS decoder then you also have an MS encoder. It's the same circuit: put LR in and you get MS out, put MS in and you get LR out. Nice...

Simmosonic wrote: I would vouch for the simplicity of a transformer-based solution, but there is one thing to keep in mind - you can't pass phantom power through a transformer.

If you actually look at the Sowter link, you'll see that their 4-transformer decoder is specifically designed to pass the phantom power through to the mics.

BTW, if you have an MS decoder then you also have an MS encoder. It's the same circuit: put LR in and you get MS out, put MS in and you get LR out. Nice...

If you ignore the 6dB level increase, yes, but you can't do the variable width trick if you only have the one matrix.

Is there a way to moniter 2 ms pairs simultaneously
without processing them or degrading the signal?

aracu wrote: Is there a way to moniter 2 ms pairs simultaneously
without processing them or degrading the signal?

I'm not sure I understand what you're getting at. To listen to M-S encoded channels, they need to be put through a decoder to give L and R signals which are then mixed into the main stereo bus. This can apply to any number of pairs, but each pair should have its own decoder. In principle, you could sum all the M channels and all the S channels and then use one decoder, but keeping track of the individual channel levels would be difficult. As far as "degrading the signal" is concerned, the M-S decoding uses one extra summation stage, which may or may not be audible. In a digital mixer, there is no signal degradation associated with M-S processing.

It's a matter of engineering technique whether you record the raw M-S channels and decode at mixdown, or whether you decode at record time and track the L-R decoded channels. Different engineers have their own preferences.

Sorry for not being clearer. I noticed that with the Sound Devices 744t,
which has 4 independant channels, that you can monitor one of two
stereo pairs configured to ms, without the signal being processed
to ms if you don´t want it to be, which is helpful if you want to tweak
the mid and side tracks independantly from one another in a computer
before processing them to ms. So based on that, I thought it would
interesting to experiment with 2 ms pairs, preferably if being able to
monitor both without processing them during the recording. And
preferably without having to use 2 Sound Devices recorders and
combining the headphone outputs.

aracu wrote: Sorry for not being clearer. I noticed that with the Sound Devices 744t,
which has 4 independant channels, that you can monitor one of two
stereo pairs configured to ms, without the signal being processed
to ms if you don´t want it to be, which is helpful if you want to tweak
the mid and side tracks independantly from one another in a computer
before processing them to ms. So based on that, I thought it would
interesting to experiment with 2 ms pairs, preferably if being able to
monitor both without processing them during the recording. And
preferably without having to use 2 Sound Devices recorders and
combining the headphone outputs.

I think there's some confusion here. When a cardioid (M) and fig-8 (S) microphones are set up, the outputs from them are termed an M-S pair, and these signals must be put through an M-S decoder to form L-R channels for mixing or monitoring (listening to). You can record either the raw M-S signals for later decoding or the decoded signals as though they were a conventional L-R pair.

The SD 744T has two built-in M-S decoders, one switchable to operate on channels 1 & 2 as a pair, and the other in the headphone output. So you can use a single M-S pair and two other mono mics and record 4 channels of decoded signals, using conventional headphone monitoring. OR, you can have two pairs of M-S mics and record them as M-S channels, using the headphone M-S decoder for monitoring. This may be what you want to do, but I'm not clear.

I love MS recording! I'm so glad that you other guys answered this poor fellows questions. You both provided beautiful answers.

Idea difficult for most people to understand that MS recording, in a sense is a 2 into 3 into 2 recording concept. And so, that is why I will generally referred to a console MS oriented systems since people can more easily see individualized the process. Where your left (middle/mono) channel is recorded as channel 1 but played back to both left and right output channels. Then the right (side) microphone is recorded as channel 2 but then played back with the use of a "mult" or "Y" cord so that it appears on 2 console input channels. That is the "3" third matrix channel.

So that second "multed" channel is bused to the left channel without phase inversion. And the third "multed" (as in the second connection of a Y splitter) channel, with phase inversion.

So now you are actually controlling 3 faders from just 2 inputs. Those three adjustable channels give you the ability to adjust both balance and width of the stereo image, by varying either the middle mono microphone without changing the level of the other 2. Or, you could vary the 2 matrix channels while not changing the middle Mono Channel by bringing up or down the "second and third" channel side microphone.

I generally find that people don't seem to understand the transformer wiring concept, as easily as they understand a console. Of course this assumes that they know how to invert the phase on the duplicated side microphone panned right, given not many consoles but professional ones seem to have phase invert switches? Which can certainly turn into a patching nightmare for those with limited means?

Your worst patch cord nightmare!
Ms. Remy Ann David

but are you sure you mean encoder and not decoder?

no, i'm not sure at all :) i did mean decoder when i said encoder, thanks for clarifying that...

i'm saving up for a two channel location rig (so as to get some relief from my studio work) and the sound devices 702 has, of late, been topping my list... but the korg ms-1000 is looking very interesting and is relatively more affordable... only thing is that it has no m/s decoding for the headphone output (unlike the sound devices unit) and i was wondering what the workaround is... i'm looking only to monitor the decoded signal, i don't want to commit it to disk... therefore either the device i'm looking for should be able to decode the headphone output, or it should have a headphone amp, if it can do only line level... as the sowter unit takes in only microphone level signals, and anyway i wouldn't want an additional transformer in the signal path, i think it will not suit the purpose.

thanks to everybody for the inputs so far.

jai shankar.

Boswell wrote: If you actually look at the Sowter link, you'll see that their 4-transformer decoder is specifically designed to pass the phantom power through to the mics.

Ah, yes! I just actually looked at the Sowter link, nice...

...but you can't do the variable width trick if you only have the one matrix.

Certainly not if you intend to use the decoder at mic level on the inputs of a mic preamplifier (unless the transformer matrix had some taps or other means to change the individual levels of the M and S input signals). But there's no problem if you do the decoding after the preamp, provided you can control the level of the individual input signals. In other words, pass the undecoded (i.e. encoded) signal through the preamp first, then into the decoder. The preamp's gain or output level controls can be used to alter the M/S ratio and therefore the width...

rfreez wrote: i'm saving up for a two channel location rig (so as to get some relief from my studio work) and the sound devices 702 has, of late, been topping my list... but the korg ms-1000 is looking very interesting and is relatively more affordable...

Save longer and get the SD:

1) It does what you want, the Korg doesn't.

2) It will retain its resale value long after the Korg has been superseded, discontinued, junked, recycled, paperweighted and laughed out of existence in time for NAMM 2008.

3) SD make serious tools for sound recordists, Korg make serious toys for musicians.

4) See (3) above.

5) Do you really want to be stringing another box of some kind (i.e. headphone amplifier with MS decoding) to the output of your recorder just so you can decode MS? More batteries, more fiddly cabling, less reliability, and more costs (short term and long term) to eat into the initial savings of the Korg.

6) See (3) above.

7) Items (1) to (6) above are the opinion of this poster only...

See (3) above.

hey simmo!

that was funny :) and i agree that the price to features ratio plus the rather "consumer gear" appearance of the korg unit causes a lot of suspicion. But then, your nagra-d costs several times as much as the SD unit, which specs out pretty much the same (including accolades, road worthiness and sound ?)... Being where i am, i'd like to think that emerging technology can bring down the cost of good gear, tho' i'd agree that theres a lot going on that suggests that my optimism is unwarranted for (behringer anybody?).

About the korg, a guy called Gus Skinas from Super Audio Center (Sonoma) shows up during the following discussion, and some pretty good things are being said about the sound quality of the Korg units, by (obviously biased, but still worth a read) beta testers.

in any case, i would never put my hard saved third world money on something that is not proven... so the korg is only a consideration at this time...

respect,

rfreez wrote: hey simmo!

that was funny :)

Funny? FUNNY?!?! I was deadly serious.

No, not really... These days I just can't keep my tongue out of my cheek.

rfreez wrote: But then, your nagra-d costs several times as much as the SD unit, which specs out pretty much the same (including accolades, road worthiness and sound ?)...

I have the Nagra V. The D was hellishly expensive, used open reel tape and weighed quite a bit more. Reputedly, the Nagra D had better mic pres and AD conversion, but the V sounds fine to me...

There is no doubt that the SD specs out similarly; that's why I referred to it as a 'serious tool for sound recordists'. There's an awful lot of crap out there, and a lot of awful crap out there too, but I don't put the SD products in that category at all. There's a sound quality threshold that separates the toys from the tools, in my opinion, and the SD products sit on the tool side of that threshold. When you cross that threshold, debates about sound quality usually revolve around subjective preferences, rather than objective measurements.

If I lost my Nagra V, I'd be tempted to replace it with a SD if for no other reason than the weight advantage. But I don't think I ultimately would because I really appreciate the Nagra V's interface, with all the important controls that I really want rapid access to (e.g. mic gain, input sensitivity, high pass filters, mono compatibility, peak hold and so on) brought out to individual knobs and switches on the front panel so I don't have to dive into menus to make a quick change. I believe that is one of the compromises you make with the SD products - much more of the control is found in the menus. That is better than compromising on the audio quality, in my books...

On the other hand, all those front panel controls make the device bigger and heavier than putting them into menus! But for the type of spontaneous on-the-spot field recordings I like to do while travelling/trekking, instant access to all of those things is essential.

rfreez wrote: Being where i am, i'd like to think that emerging technology can bring down the cost of good gear, tho' i'd agree that theres a lot going on that suggests that my optimism is unwarranted for (behringer anybody?).

But there's also a lot going on that warrants that optimism. The question is, how long do you want to wait for it to arrive? And what compromises are you prepared to accept?

Chip-based AD converters, for example, have made considerable progress over the last 10 years, and that is one of the areas that used to be very expensive to do well.

rfreez wrote: in any case, i would never put my hard saved third world money on something that is not proven... so the korg is only a consideration at this time...

It is also a matter of religion. Either you believe in PCM or you believe in DSD - have you noticed how polarised that subject becomes? Personally, I think that DSD sounds wonderful, but all those who think it is going to replace PCM are seriously deluded. Its window of opportunity came and went a long time ago - they threw it up the flagpole with much ado, and the market stayed away from it in droves. It will probably limp along on the life-support of a dedicated bunch of evangelists for years to come, rather like MiniDisc, Commodore 64s, digital tape-based formats, GW Bush and other things that simply refuse to go away and leave us alone, no matter how much we ignore them.

The interesting question is this: why did Korg, of all companies, choose DSD for such a device? What kind of deal was struck with Sony to make that happen? I doubt Sony would want to put a low-cost DSD recorder to market under their own name, it would soil the image they've tried to create for DSD. But encouraging a manufacturer like Korg to do it would be a clever move. Sony have found over and over again that their ingeniously-conceived consumer formats often fail in the consumer market but eventually find a home elsewhere. It happened to Beta video tape, it happened to DAT, and it happened to MD (initially).

Where will DSD go? Toys for musicians from Korg???

A couple of late entries. Since I haven't read all of the replies (party foul), this may be out of place... but here goes.
http://www.peterengh.com
http://www.dangerousmusic.com/s&m.html

Hope that helps.

Re: m/s encoder

First let me state that I feel m/s encoders and decoders are additional un-wanted signal processors in a recording chain. But If one is committed to using m/s technique, I feel that the available software solutions are more preferable. From what I have gathered from your posts here, the cost of building a really transparent analog decoder will be too high for your budget. This cost would be better spent on other things.

I picked up a used AEA MS38 Lite on Ebay for around $100. Very handy as a decent headphone amp with MS decoding. I also made two channels of transformer-based (Sowter) headphone decoders and stuck them in some extra space in a rack chassis that had 4 channels of mic splitter. Both work fine for field monitoring. The Sowter transformers were around $35-$40 each, I think.

I agree that for final decode, use software. But these other things are handy for monitoring (at least before I got my SD 744T, which has built-in MS decoding on the headphone out).

Michael

thanks schizojames... the peter engh device seems to be the closest commercially available portable device, but the problem of headphone monitoring remains.

I agree that for final decode, use software. But these other things are handy for monitoring

exactly!

I also made two channels of transformer-based (Sowter) headphone decoders

is it the Type 8392?

I know NOTHING about circuits and electronics, but would the following (obviously basic) schematic from Sowter be clear to most novice gear servicing engineers...? to me it looks like all it needs is a couple of female 1/4" sockets, a chasis, the tranny and the 15 ohm resistor... is it as simple as this?

thanks,

Yes, that circuit works for headphones. If you are going to wire it in series with the existing headphone output, you need to build it with a DPDT switch to take it out of circuit when monitoring conventional L-R signals. You would engage the decoder only when the pair of channels you were monitoring were M-S encoded but panned L and R.

Note that this would not normally work for soloed channels from a mixer where the solo monitoring is the centred mono sum of the selected solo channels.

You would need to check how the headphone monitoring works on the Korg MR-1000 (assuming you meant MR-1000 and not MS-1000 that you typed). The unit has 2-channel recording and playback capability, where obviously the two mic inputs go to separate channels, but it doesn't say for definite in the manual that the headphone output is stereo L-R when monitoring during recording. I assume it is, but I've been caught out in the past by making assumptions about the way equipment works.

Re: above schematic

You may need to take care with what that circuit is connected to. The following device will be loading that network. Transformers are reactive components. That means that it will react differently depending on signal frequency. Loading (terminating) the transformer with a proper Z will optimize linear response in the audio band.

I don't care much for this circuit because it has no control of mid-side balance. To me, that is the whole advantage of the technique: flexibility. If all your going to do is decode to x-y, why not just use x-y technique in the first place?

Zilla wrote: If all your going to do is decode to x-y, why not just use x-y technique in the first place?

Because sound hits your mikes on-axis with M/S. This is one of the most important advantages of the technique apart from post processing of width. The tonal improvements are significant.

DavidSpearritt wrote: Because sound hits your mikes on-axis with M/S. ... tonal improvements are significant.

The sound source can only be on-axis with one mic (M), and only when the primary sound source is dead in the center (ex: a soloist). True, there can be a tonal improvement in some situations, and hopefully one that overcomes the potential coloring of the decoding process. For fun, one could argue that with larger ensembles an x-y array has more information on-axis than a m/s (ex: center main pair of an orchestra 4-way). But making such an arguement would only peg me as an audio nerd, so l will skip it.

Zilla wrote: If all your going to do is decode to x-y, why not just use x-y technique in the first place?

M-S sounds different from X-Y on most sources. Not always better, but different. It is another option in the recording engineer's array of options. But it's surprising how often you turn to M-S when faced with a "now how shall I try miking this up?" situation.

You may need to take care with what that circuit is connected to. The following device will be loading that network. Transformers are reactive components. That means that it will react differently depending on signal frequency. Loading (terminating) the transformer with a proper Z will optimize linear response in the audio band.

This is specifically a headphone monitoring decoder circuit. The specified transformer is designed for headphone power levels. It's driven from a headphone amplifier and is loaded by the impedance of the headphones. The 15 Ohm resistor compensates the M-channel for the resistive losses of the transformer primary in the S-channel.

I don't care much for this circuit because it has no control of mid-side balance. To me, that is the whole advantage of the technique: flexibility.

This circuit is for monitoring M-S encoded channels as they are being recorded still in their encoded form. You adjust the M-S balance (and hence the L-R width) at mixdown. At that stage, you use a separate decoding matrix, not this headphone decoder, between the M-S inputs and the L-R stereo mix bus. For a two-channel recording, the mixdown may simply consist of M-S level adjust followed by decode to L-R, but that is just a trivial case of a multi-track recording.

Yes, that's the transformer I used. When I got them I had to wait for shipment from UK, but the US importer told me he would stock a few.

The circuit is that simple and, in my experience with a variety of headphones and driving sources, works fine.

Regarding M/S balance, you just adjust the feeds to the transformer. Works fine.

Michael

Boswell wrote: M-S sounds different from X-Y on most sources. Not always better, but different. It is another option in the recording engineer's array of options.

Fair enough.

Boswell wrote: The specified transformer is designed for headphone power levels. It's driven from a headphone amplifier and is loaded by the impedance of the headphones.

Headphone Z does vary across makes and models. A 30ohm HP will load those xfrmrs differently than a 600ohm pair. It would be interesting to see if it made any appreciable difference in response. I would imagine that the manufacture's design would have made allowances for this.

Boswell wrote: This circuit is for monitoring M-S encoded channels as they are being recorded ... adjust the M-S balance...at mixdown.

Yes, this would function and get you through the session. For me, it is not an ideal solution. Being able to monitor the decoded signal better informs the engineer and helps optimize mic placement. It allows the engineer to 'play' with the interaction between m/s width and mic placement. I guess I am just uncomfortable with waiting for post-production to hear the what the possibilities might be.

mdemeyer wrote:
Regarding M/S balance, you just adjust the feeds to the transformer. Works fine.

Boy, that was simple! I feel silly now.

sorry to go off topic, but,

from the first page of the mr-1000 instruction manual:

Using the unit in the following locations can result in a malfunction.

• In direct sunlight
• Locations of extreme temperature or humidity
• Excessively dusty or dirty
• Locations of excessive vibration
• Close to magnetic fields

is this a location recorder or what? :roll:

this is what the sound devices manual says:

The 702 output meter is unaffected by shock or extremes in temperature and humidity.

and either the instruction manual is woefully incomplete or the damn thing doesn't even allow you to adjust the balance between, and levels of the two inputs, for the headphone output :shock:

what simmo said about the unit being a toy is becoming more and more apparent...

rfreez wrote: either the instruction manual is woefully incomplete or the damn thing doesn't even allow you to adjust the balance between, and levels of the two inputs, for the headphone output :shock:

Yes, I think the headphone monitor feels a bit of a bolt-on. It's got a control marked "volume" and nothing else. Bear in mind, though, that the unit is basically a two-track recorder, and the live headphone monitoring is a check that you are recording something on each of the two tracks, not an emulation of studio monitoring during post processing.

Given the necessary limitations of the unit, I think the transformer M-S decoder for the headphone channel is a good solution, affording you the choice of L-R (in its various forms) or M-S miking in the field. You could house the transformer in a convenient lump in an adaptor cord along with a TRS jack socket and a by-pass switch. The other end of the cord would have the TRS jack plug to mate with the regular headphone output on the MR-1000. The switch avoids the need to keep plugging and unplugging the adaptor cable when listening to replays of L-R and M-S recorded tracks.

Zilla wrote: [quote=DavidSpearritt]Because sound hits your mikes on-axis with M/S. ... tonal improvements are significant.

The sound source can only be on-axis with one mic (M), and only when the primary sound source is dead in the center (ex: a soloist). True, there can be a tonal improvement in some situations, and hopefully one that overcomes the potential coloring of the decoding process. For fun, one could argue that with larger ensembles an x-y array has more information on-axis than a m/s (ex: center main pair of an orchestra 4-way). But making such an arguement would only peg me as an audio nerd, so l will skip it.

Two counter points, a fig 8 has much better off axis tonality than a cardioid, so the side mic sounds much better than another card, giving both mics of an MS pair significantly better tonal response to any off axis sounds.

MS shouldn't and isn't used for big sources by those in the know. Its best for small ensembles that fit into about 90 deg or less. For large sources a couple of omnis starts to work well.

Two good points. Generally bi-directional mics do have smoother off-axis response. The question is does a given bi-directional mic have an over-all preferable sound than a given cardioid. The choice is completely dependent on situation and taste.

I agree with you that coincident mic technics (x-y, m/s) are most appropriate for small ensembles. However, there are a number of professional engineers out there who indeed employee m/s on orchestra's, opera and other large ensembles. The sonics are not my cup of tea, but the record sales obviously show that it can be successful.

Boswell wrote: Given the necessary limitations of the unit, I think the transformer M-S decoder for the headphone channel is a good solution, affording you the choice of L-R (in its various forms) or M-S miking in the field.

There may be a problem here that we have not considered because the Sowter, like other transformer decoders, is a passive device and therefore will incur an insertion loss of power. This could be a bad thing when driving it from the headphone output of the Korg, which is ultimately a low-voltage battery-powered device and can therefore only produce a relatively small output voltage swing (compared to a mains-powered device with supply rails of +/-15V or more). To get more power into the headphones requires drawing more output current, and that in turn requires lower impedance headphones.

This is particularly important in situations where it is necessary to monitor the recording while in the same space as the performers, because, with such a decoding circuit in place, it may be impossible to drive the headphones to a sufficient SPL for meaningful monitoring without clipping.

Although I was one of the supporters of the transformer solution, I was not thinking of the Korg's headphone driving capability at the time. I suspect an active device that decodes MS and drives headphones may be more appropriate. Or better yet, ditch the Korg and go with the SD!

"The SD 744T has two built-in M-S decoders, one switchable to operate on channels 1 & 2 as a pair, and the other in the headphone output. So you can use a single M-S pair and two other mono mics and record 4 channels of decoded signals, using conventional headphone monitoring. OR, you can have two pairs of M-S mics and record them as M-S channels, using the headphone M-S decoder for monitoring. This may be what you want to do, but I'm not clear."

Boswell, thanks for taking the effort to answer my question.
I noticed that with the 744t, you can monitor an ms pair through
the headphone output without processing the signal. So what I
was wondering was if there were a cost effective way of monitering
2 ms pairs without processing (those 4 channels divided into two
separate pairs of ms). An expensive way to do it would be to use
2 Sound Devices recorders and combine the headphone outputs
to stereo. It's possible that there isn't a simple way of doing it due
of the nature of standard ms matrixes.

Reme, I am not impressed by your condescending attitude,
which has already been commented on in these forums.

aracu wrote: "The SD 744T has two built-in M-S decoders, one switchable to operate on channels 1 & 2 as a pair, and the other in the headphone output. So you can use a single M-S pair and two other mono mics and record 4 channels of decoded signals, using conventional headphone monitoring. OR, you can have two pairs of M-S mics and record them as M-S channels, using the headphone M-S decoder for monitoring. This may be what you want to do, but I'm not clear."

Boswell, thanks for taking the effort to answer my question.
I noticed that with the 744t, you can monitor an ms pair through
the headphone output without processing the signal. So what I
was wondering was if there were a cost effective way of monitering
2 ms pairs without processing (those 4 channels divided into two
separate pairs of ms). An expensive way to do it would be to use
2 Sound Devices recorders and combine the headphone outputs
to stereo. It's possible that there isn't a simple way of doing it due
of the nature of standard ms matrixes.

I haven't got the 744T manual to hand, but my memory of how the headphone monitor works is that you have 20+ options to choose from, including having two M-S pairs as the four channels and alternating the monitoring between the two pairs. If you want to hear the summed L and R from the decoded M-S pairs, I think you're out of luck using the options available on the machine. However, if you were to select "Monitor 1+3, 2+4", you could get encoded M-S summed channels at the headphone jack, so an external transformer-based MS decoder would give you an L-R pair. Quite what that would mean as a sound field would depend entirely on your microphone placement, but presumably that would be your intention for post processing anyway.